The present invention relates generally to digital audio broadcasting (DAB) and other techniques for transmitting information in a communication system.
Proposed systems for providing digital audio broadcasting (DAB) in the FM radio band are expected to provide near CD-quality audio, data services, and more robust coverage than existing analog FM transmissions. However, until such time as a transition to all-digital DAB can be achieved, broadcasters require an intermediate solution in which the analog and digital signals can be transmitted simultaneously within the same licensed band. Such systems are typically referred to as hybrid, in-band on-channel (HIBOC) DAB systems, and are being developed for both the FM and AM radio bands. In order to prevent significant distortion in conventional analog FM receivers, the digital signal in a typical FM HIBOC DAB system is, e.g., transmitted in two sidebands, one on either side of the analog FM host signal.
Perceptual audio coding techniques are particularly attractive for FM band and AM band transmission applications such as HIBOC DAB. Perceptual audio coding devices, such as the perceptual audio coder (PAC) described in D. Sinha, J. D. Johnston, S. Dorward and S. R. Quackenbush, xe2x80x9cThe Perceptual Audio Coder,xe2x80x9d in Digital Audio, Section 42, pp. 42-1 to 42-18, CRC Press, 1998, which is incorporated by reference herein, perform audio coding using anoise allocation strategy whereby for each audio frame the bit requirement is computed based on a psychoacoustic model. PACs and other audio coding devices incorporating similar compression techniques are inherently packet-oriented, i.e., audio information for a fixed interval (frame) of time is represented by a variable bit length packet. Each packet includes certain control information followed by a quantized spectral/subband description of the audio frame. For stereo signals, the packet may contain the spectral description of two or more audio channels separately or differentially, as a center channel and side channels (e.g., a left channel and a right channel).
PAC encoding as described in the above-cited reference may be viewed as a perceptually-driven adaptive filter bank or transform coding algorithm. It incorporates advanced signal processing and psychoacoustic modeling techniques to achieve a high level of signal compression. In brief, PAC encoding uses a signal adaptive switched filter bank which switches between a Modified Discrete Cosine Transform (MDCT) and a wavelet transform to obtain: compact description of the audio signal. The filter bank output is quantized using non-uniform vector quantizers. For the purpose of quantization, the filter bank outputs are grouped into so-called xe2x80x9ccoderbandsxe2x80x9d so that quantizer parameters, e.g., quantizer step sizes, are independently chosen for each coderband. These step sizes are generated in accordance with a psychoacoustic model. Quantized coefficients are further compressed using an adaptive Huffman coding technique. PAC employs a total of 15 different codebooks, and for each codeband, the best codebook may be chosen independently. For stereo and multichannel audio material, sum/difference or other form of multichannel combinations may be encoded.
PAC encoding formats the compressed audio information into a packetized bitstream using a block sampling algorithm. At a 44.1 kHz sampling rate, each packet corresponds to 1024 input samples from each channel, regardless of the number of channels. The Huffman encoded filter bank outputs, codebook selection, quantizers and channel combination information for one 1024 sample block are arranged in a single packet. Although the size of the packet corresponding to each 1024 input audio sample block is variable, a long-term constant average packet length may be maintained as will be described below.
Depending on the application, various additional information may be added to the first frame or to every frame. For unreliable transmission channels, such as those in DAB applications, a header is added to each frame. This header contains critical PAC packet synchronization information for error recovery and may also contain other useful information such as sample rate, transmission bit rate, audio coding modes, etc. The critical control information is further protected by repeating it in two consecutive packets.
It is clear from the above description that the PAC bit demand is derived primarily by the quantizer step sizes, as determined in accordance with the psychoacoustic model. However, due to the use of Huffman coding, it is generally not possible to predict the precise bit demand in advance, i.e., prior to the quantization and Huffman coding steps, and the bit demand varies from frame to frame. Conventional PAC encoders therefore utilize a buffering mechanism and a rate loop to meet long-term bit rate constraints. The size of the buffer in the buffering mechanism is determined by the allowable system delay.
In conventional PAC bit allocation, the encoder makes a request for allocating a certain number of bits for a particular audio frame to a buffer control mechanism. Depending upon the state of the buffer and the average bit rate, the buffer control mechanism then returns the maximum number of bits which can actually be allocated to the current frame. It should be noted that this bit assignment can be significantly lower than the initial bit allocation request. This indicates that it is not possible to encode the current frame at an accuracy level for perceptually transparent coding, i.e., as implied by the initial psychoacoustic model step sizes. It is the function of the rate loop to adjust the step sizes so that bit demand with the modified step sizes is below, and close to, the actual bit allocation. The rate loop operates based on psychoacoustic principles to minimize the perception of excess noise.
Despite the above-described advances in DAB systems which utilize PAC encoding, a need exists for further improvements in techniques for transmitting digital audio and other information, so as to provide enhanced performance capabilities in these and other systems.
The present invention provides methods and apparatus for implementing error screening in digital audio broadcasting (DAB) systems or other types of digital communication systems, so as to provide enhanced performance relative to conventional systems.
In accordance with the invention, control information associated with a given packet of the received information is identified and compared with a decoding requirement of the packet, in order to control the generation of an error indicator for the packet. The error indicator may be generated in response to an inconsistency between the control information and the decoding requirement. For example, the control information may include an indication of packet length that can be compared to a number of bits required in a Huffman decoding process applied to the corresponding packet, with any inconsistency between the packet length indication and the number of required bits leading to the generation of an error flag for the packet.
The digital information may be encoded compressed audio information in the form of a bitstream including a series of packets generated by a perceptual audio coder (PAC) encoder or other suitable encoder. Error flags generated as a result of the error screening may be utilized to trigger an error mitigation algorithm in a PAC decoder. As another example, the error flags may be supplied to a channel decoder and used in the selection of alternative decoding paths, in accordance with a List Viterbi algorithm technique.
Although particularly well-suited for detecting bursty channel errors, the above-described error screening can detect any type of transmission error, and does not require any particular type of transmission coding. In addition, error screening of this type can be used in conjunction with other types of error detection, e.g., cyclic redundancy code (CRC) error detection. For example, error screening based on Huffman code and control information consistency may be used in conjunction with outer code CRC error detection in order to achieve a desired level of performance with a less powerful CRC.
Moreover, although illustrated herein using an embodiment which includes both an inner channel code and an outer channel code, the invention can be implemented with either an inner code or an outer code, or with no channel code.
The invention can be applied to other types of digital information, including, for example, data, video and image information. In addition, the invention may be implemented in numerous applications other than FM and AM hybrid, in-band on-channel (HIBOC) DAB systems, such as Internet and satellite broadcasting systems, systems for simultaneous delivery of audio and data, etc. Moreover, the invention is applicable not only to perceptual coders but also to other types of source encoders using other compression techniques over a wide range of bit rates.